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How many bits, how fast, just how much resolution is enough?


BlueSkyy

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Well in my case because I took my own advice, and stopped paying attention when I saw the stair case looking graph. Also because you mentioned the more points you have taken the more accurate. That describes a higher vs lower sample rate.

 

Just a kind of picky point, but that depends (more points describing a higher vs. lower sample rate).

 

Sample rate and word depth are related (obviously). Interpolation in time (sample rate) does no good unless you can also interpolate in amplitude (bit depth), *unless* you aren't in the PCM world, but the SDM/PDM world, where each sample value stands not for an absolute amplitude but a relative one (i.e., each value doesn't independently determine amplitude, rather amplitude depends upon the sequence of values).

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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I can agree with that... in a lot of systems it's probably more about DACs and how they filter than data depth / rate (after Redbook).

 

Redbook sounds great to me, I don't personally see the need to go with anything higher in my system.

But your system does. From a review of the Vega:

 

Yet the Vega is their most ambitious launch yet. As does Wadia for their Intuition 01, today's DAC/preamp—AURALiC prefers digital processor—upsamples all PCM to 1.5MHz and 32-bit depth.

 

The Auralic uses an ESS SABRE DAC chip to do this, foregoing ESS's internal upsampling to ~44.1MHz.

 

That's why these discussions always strike me as such a waste of time. Just about no one in these forums is actually listening to analog that's been reconstructed straight from RedBook. The vast majority are listening to analog reconstructed from a bitstream running at somewhere between 2.8MHz and 11.2MHz. A substantial minority of folks whose DACs use the ESS chip's internal upsampling are listening to analog reconstructed from a bitstream running at about 44.1MHz.

 

The engineers who built your DACs decided long ago (about 25 years - that's when delta-sigma DACs began to take over the market) that MHz sample rates were the least expensive most practical solution. If you want to talk about theory rather than reality, that was also resolved very long ago (implied by Nyquist's work in 1928, with Shannon's mathematical proof published in 1948). A sample rate just over twice the highest "frequency of interest" is all you need to satisfy the conditions of the mathematical proof.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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where can I read your published comments? I assume you are a Fellow of the AES, right?

Companies selling high resolution recordings commercially have used Miska's sigma delta modulators in preference to any others on the market for demonstration quality offerings. Roon, expensive and sophisticated GUI player software for audiophiles, chose Miska's oversampling software, including his filters and modulators, to ally themselves with, in preference to developing their own, and they are an audio software company. If Miska is not a member of AES (I don't know whether he is or not), or not published (again, I don't know whether he is or not), then perhaps these audiophile recording and software companies are just stupid for hiring a pretender without the proper credentials, eh?

 

Or perhaps they know something; perhaps Miska knows something; and if one asks instead of challenging, perhaps it is possible to learn something.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Any that supports and promotes his commercial objectives.

 

That's why I like reviewing the history of CD players, DACs, and RedBook audio. Within just a couple of years after CDs and CD players were first sold, 8x oversampling had become the standard for DAC chips. These were commodity chips, not specialist audiophile pieces. Within just a few more years, sigma delta modulation to MHz sample rates had become the standard, again in the commodity audio chip market. Before 8x oversampling became commonplace, CD players of the day (and NOS players and NOS DACs still today) fed at RedBook rates showed easily measurable harmonic and intermodulation distortion due to aliasing.

 

Then the first pro/audiophile asynchronous sample rate conversion (ASRC) DACs hit the market, most notably from Benchmark and Lavry. These DACs resampled to non-even multiples of the input rate (in Benchmark's case, 110 KHz; I don't know what Lavry used, though I recall the input rate was limited to 96KHz) as a means of jitter reduction, prior to putting the resulting bitstream through sigma-delta modulation to MHz sample rates. I recall Lavry in particular publishing white papers saying one certainly didn't need higher sample rates than 96KHz and suggesting higher rates were actually deleterious, while not mentioning their own DACs internally used the same MHz sample rates as everyone else.

 

But I'm sure these papers, which did as much as anything to spur discussion of how 44.1 or surely 96KHz rates were perfectly adequate (when even inexpensive non-audiophile commodity players used chips designed by non-audiophile engineers that upsampled internally to MHz rates) had nothing to do with Lavry's commercial objectives. :)

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Meyer, E. B. and D. R. Moran. 2007. Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback. JAES 55(9): September 2007

 

Abstract:

Claims both published and anecdotal are regularly made for audibly superior sound quality for two-channel audio encoded with longer word lengths and/or at higher sampling rates than the 16-bit/44.1-kHz CD standard. The authors report on a series of double-blind tests comparing the analog output of high-resolution players playing high-resolution recordings with the same signal passed through a 16-bit/44.1-kHz "bottleneck." The tests were conducted for over a year using different systems and a variety of subjects. The systems included expensive professional monitors and one high-end system with electrostatic loudspeakers and expensive components and cables. The subjects included professional recording engineers, students in a university recording program, and dedicated audiophiles. The test results show that the CD-quality A/D/A loop was undetectable at normal-to-loud listening levels, by any of the subjects, on any of the playback systems. The noise of the CD-quality loop was audible only at very elevated levels.

The CD recorder used for the Meyer and Moran tests was the HHB CDR-850, which used typical delta-sigma chips. So the test design identified (at best) whether people could blindly identify a signal run through an extra KHz-to-MHz step before conversion to analog, rather than any "pure" comparison of different sample rates.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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There are dozens, likely hundreds. of published studies showing lack of statistical differences for the higher bit rates. Maybe they are all wrong? Personally, it doesn't cost me a lot to get that source material, but I figure I am likely getting more care in recording/mastering...

Similarly, there some studies showing that mp3 files cannot be distinguished from redbook. It costs me nothing to use Apple Lossless instead tho (well, ok - I guess it cost me a little more for a higher capacity iPhone).

There was a test done here at CA years ago by a member known as Julf, with the cooperation of the head of the Bis recording label. It included RedBook, mp3, and hi res files. The "winner" by a fairly large margin was a RedBook file 1dB louder than the others. This quite clearly shows even a loudness difference so small that it is not consciously perceptible is sufficient to sway preference in the context of an audio A/B test. (No tester identified from listening that one of the files was louder.)

 

By the way, if one was listening for a difference between a RedBook and a direct to DSD recording of the same material, what audio characteristics - phase, frequency response, time domain response, imaging, soundstage, noise - should one listen for, and what should these differences sound like?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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And your result still doesn't tell anything about audibility of the format differences alone, only tells about that particular DAC.

 

In addition, even in digital domain, there is no single way to do the conversion to/from RedBook, each of those also produce different results.

 

So overall this is complex topic and the results are not straightforward.

 

 

Word (as they say). ;)

 

 

As a lawyer, what I nearly always find when non-lawyers talk about the law is that they tend to leap to sweeping over-generalized conclusions. I'm sure something very similar occurs to you as you read what non-professionals write about filters, modulators, audio software, etc.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Not so sure about that... the question is whether 16/44 is good enough as source data to properly deliver the content within the hearing range of humans (frequency & dynamic range) on DACs than are designed well enough to let that happen.

 

"How many bits, how fast, just how much resolution is enough?"

 

... if one of his DACs tells him redbook is enough, then his post is right on topic in my book. If his other DACs don't, that's not necessarily redbook's fault.

 

 

---on a another note---

 

If we (CA) agree that freq range 20hz-20khz is being captured and reproduced, then dynamic range shortfall of 16 bit is next to examine, which is fine on paper but not much music contains huge dynamic ranges like that... DR Database for example makes for some depressing reading.

 

 

;-)

 

 

Hi r_w, been meaning to get back to you regarding your earlier response and why I talked about what I did (the history of CD players, DACs, oversampling, filtering, etc.).

 

Frequency response is really not in question here. *It is the wrong thing to be looking at when discussing the adequacy of digital audio standards.* Why is that?

 

- First, it's a fact that a 44.1KHz sample rate is mathematically adequate to reproduce all audible frequencies. If you want a way to picture what's going on with the math, here's one:

 

Take a single sample point. There are an infinite number of curves that can be drawn through it.

 

Add a sample point, for a total of two. There are a smaller number of curves that will run through both points.

 

Now adjust your sample rate to just above double the highest frequency rate you are interested in. That adds a third sample point. As long as we are talking about frequencies that don't exceed just under half that sampling rate, there is one and only one curve that will run through all three sample points. So you have defined the sampled curve along its entire length and not just at the sampled points. That's a simplified English version of what the Sampling Theorem says. That's why more samples will not help to define the signal better. So a 44.1KHz sample rate will perfectly reproduce any signal of just under 22.05KHz or lower.

 

- Well then why upsample?

 

The problem is not the sample rate itself and its adequacy for reproducing the frequency range - as we've already seen, we can perfectly define any signal within our frequency range of interest with a 44.1KHz sample rate. The problem is that the Sampling Theorem requires a perfect filter that will pass all frequencies below half the sample rate, and completely eliminate all frequencies from half the sample rate on up. This perfect filter would have to operate instantaneously, in no time at all, would have to keep operating over an infinite period of time, and would have to operate perfectly as to frequency, i.e., would have to create an absolutely perfect vertical line between full signal and no signal on an oscilloscope frequency sweep. No such animal exists.

 

Because there are no perfect filters, all filters must balance two types of errors, frequency domain errors (aliasing or "leaking") and time domain errors ("ringing" or "smearing"). Mathematically, improving your filter's behavior in one aspect worsens it in the other aspect - that is, for a given filter, adjusting its parameters for less aliasing gives you more ringing, and vice versa. That's just sheer math, and it's unavoidable. So what you need to be looking at when discussing the adequacy of digital audio standards (such as sample rates) is not frequency response, but aliasing and ringing distortion.

 

The entire reason for upsampling is to make it more practical (easier and cheaper) to design filters with adequate distortion performance.

 

OK then, how should we identify adequate performance? What should we listen for?

 

Well, this comment is already long enough. Perhaps in my next one.... ;)

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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You also should do the same comparison using HQP and/or AuI to upconvert to the native format of your DAC from these source files.

 

This will test whether the differences you hear are related to the filtering/sampling done by the DAC (vs capabilities of CPU/software)

 

Yes, and not incidentally it will get around two tremendous problems with such comparisons - they often involve different masterings and/or differences in loudness between the two files.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Indeed.

There are so many variables: upsample to PCM or DSD, to a multiple frequency or maximum DAC sample rate, minimum phase or linear phase filters, SOX vs izotope vs HQPlayer vs Aul vs DAC ASRC, in line or off line, etc.

 

This is what I dislike about computer audio, it's as fussy as vinyl...

 

R

 

You can certainly choose a "plug and play" experience that can be very, very good. But there are folks (counting myself among them, or us) who would like to try to get the very best possible from whatever we have or can reasonably afford, and at least at this point in the development of digital audio that does involve some fuss.

 

You have your choice of whether the fuss occurs on the learning end or on the playback end, i.e., whether to try to learn about the reasons behind the way things sound or spend time on trial and error with playback settings.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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early CDs were notorious for bad mastering; that is likely what you heard

 

a fellowship in a professional scientific or engineering society is a rather large chunk of cheese

 

you realize that I have asked for actual studies, rather than arguments from authority, but what the advocates of higher bitology have posted is almost nothing but arguments from authority ???

 

We certainly know it becomes easier/cheaper to achieve filters with levels of distortion acceptable to engineers at higher sample rate input. We're then left with the question whether levels of distortion considered unacceptable by engineers make differences that are audible, or even, short of being audible, have other conscious or subconscious effects (for example, "listening fatigue").

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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this is well known and the usual spec. is 0.2 or even 0.1 dB - it has to be controlled for and usually not just by ear either

 

Yes, exactly. Do you suppose if we'd been told one of the files was 1dB louder we could have identified which one?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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If you read Lavry's papers he isn't hiding anything. He talks about how his own converters are delta sigma and run in mhz range. Even then what he talks about as getting less accurate sampling at too high a speed and fewer effective bits is still in effect. He warns the reader to keep both sampling rates clearly separate in their minds when trying to understand how this works. Doesn't sound like someone pulling a fast one if you read it rather than what others say about the paper.

 

http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

 

 

Perhaps it's just the way Lavry's writing strikes me, but on re-reading I once again feel his arguments are far more slippery than clear.

 

He implies the decimation from the original SDM processed bitstream at MHz rates to RedBook at 16 bit 44.1KHz is needed for "accuracy," when in fact it is marketing-driven by the need to convert to CDs for sale. In doing this he completely bypasses the issue of whether the distinction he repeatedly makes between "audio sample rates" (the rate of the audio data) and "other sample rates" (such as the rates used by DACs internally) needs to exist at all.

 

The paper states (at page 23 of 27) "In the case of DA converters, the data is interpolated to higher rates which help filtering and response. Such oversampling and up sampling are local processes and tradeoff aimed at optimizing the conversion hardware." [Emphasis added.]

If the original "audio sample rate" doesn't go through the decimation step but stays at the original higher rate, then one gets to "help filtering and response" and "optimiz[e] the conversion hardware" without requiring the "local processes" of oversampling/upsampling. And if these higher rates do indeed "help filtering and response," then where exactly is the tradeoff between speed and accuracy he's trying to sell?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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re: it's a fact that a 44.1KHz sample rate is mathematically adequate to reproduce all audible frequencies.

 

I would say (based on a post somewhere up above) that it's a fact that a 44.1KHz sample rate is mathematically adequate to reproduce all audible [sine wave] frequencies.

 

i.e. out to 20 kHz

 

if there is an issue with the 1930s research that established that limit, I bet that is for transients

 

Careful. I believe the Sampling Theorem works for non-periodic signals just as well. They must still obey the rule that the sampling rate must be more than double the highest "frequency of interest," which in the case of transients would require sampling at more than double the rate of change of the transients. But I am not aware of undisputed research that we can hear transients with rates of change greater than 20,000Hz. If there is such research, I'd be grateful to see it or a link to it.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Hi Jud,

 

Good post, thanks for the explanation (TBH... I get a bit lost when talk moves to impulse / ringing / etc... that's way over my pay grade).

 

 

 

Ringing has to do with something called the Gibbs Phenomenon, which I don't understand either :) , and which again has to do with math: https://en.wikipedia.org/wiki/Gibbs_phenomenon .

 

There are a couple of funky things about the ringing associated with the Gibbs Phenomenon -

 

- For the filters used in digital audio, the ringing occurs at ultrasonic frequencies, so audible effects, if any, must be in the time domain. Miska and others have described the effect as "smearing" the signal in time.

 

- Ringing occurs both before and after the main impulse as long as the filter is linear phase. Even though the notion is a bit brain-twisting, this does not violate cause and effect, but it does mean the impact of the ringing (e.g., the "smearing") will occur both leading up to and after the main impulse. Minimum phase filters move the ringing/smearing mainly or exclusively after the impulse. This is one of the reasons minimum phase filters are used, to minimize or eliminate "smearing" prior to, and thus the sonic impact of, the leading edge of sharp impulses/transients, on the theory that "smearing"/ringing on and after the trailing edge of the impulse/transient won't sound terribly unnatural.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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I'm assuming for this post that 16/44 captures all of the frequency range and transient information we need (you may disagree, but make that assumption for a moment). Then the real reason for hi-res (or upsampling) is all about the digital-to-analog conversion process and the benefits we get by moving filters and other conversion artifacts as far above 20kHz as we can.

 

That results in

a) our wanting to feed our DAC hi-res (call it DSD128 or higher or DXD) information (irregardless of whether it started as 16/44 that we upsampled or an actual DSD128 or DSD512 or DXD file);

b) then applying less aggressive filters (so as to avoid ringing and other artifacts in the 20Hz-20kHz range only); and

c) feeding all equipment downstream of the DAC considerable content above 20kHz (unless the DAC itself introduces a 20kHz cliff filter).

 

If we are doing a,b and c, then how important is it for us to know how each piece of downstream equipment will interact with those higher than 20kHz frequencies (i.e. pre-amp, amp and tweeters)?

 

Some have made the case for having pre-amps, amps and speakers that are linear out to 40kHz or more such that feeding them information not brickwalled at 20kHz doesn't introduce new problems that we were trying to solve with the hi-res and gentle filters solution at the DAC. Is it therefore possible that the reason some of us do/don't prefer (or hear) the benefits of hi-res is the limitations of the downstream equipment?

 

 

You're missing one piece, which is that moving the sample rate higher permits exclusion of near-20KHz ultrasonics without a "brickwall" filter. For DSD, ultrasonic noise is moved further from the audible range.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Well this was written in 2004

 

But as you point out in a part of your post I snipped, he said the same in 2012, and as far as I know continues to maintain the same position. See, e.g., http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf

 

The idea to keep pushing sample rates up both in the operation of delta-sigma designs and in the PCM or DSD available rates is an easy selling point. Just like megapixels for cameras. I would be more convinced if someone had rational explanations for what point it sounds so good no improvement can be detected with higher sample rates. So far it is just higher is better as if the improvement can be extended forever with more more more.

Yes, we will always have specsmanship in the marketing of these products. On the other hand, I don't see *current* support for Lavry's position that anything over a 96KHz sample rate leads to worse results.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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I'm not sure this from Miska -

 

As long as measurable (and/or audible) signal fidelity keeps improving by improving implementation, I will keep doing it ad infinitum. I have no reason not to. So far I've been also hearing improvements, but not going to claim anything about anybody else's hearing or non hearing. That's something they need to decide on their own.
[emphasis added]

 

- really deserved the rather negative responses to it. Seems pretty reasonable to me.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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God help us all if the culinary world develops the same audiophobia criteria where everything matters! Who wants a lab technician instead of a Michelin chef?

The chef/owner of what was by many accounts the best restaurant in the world for years would likely disagree with you. Ferran AdriÃ*'s ElBulli Lab in Barcelona - Bon Appetit | Bon Appetit

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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There is no such thing as two simultaneous tones of the same frequency.

 

Don't tell that to the orchestra! :)

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Of course there can be... :D

 

Theorems don't prove anything about hearing, only maths that happen in digital domain and only under certain conditions.

I would say in this case they would indicate where it would be best to look for any advantage of hi res, and that would not appear to be with better tracking of the signal but with the opportunity for better filtering.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Instruments don't produce pure sine waves.

 

True - they produce *combinations* of pure sine waves.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Your figures prove that the 44.1 kHz sampled signal contains everything needed for accurate reconstruction. That incorporating a digital processing step aids in producing a quality output is completely beside the point.

It's beside the point you are making (that more samples aren't needed to accurately represent the waveform), but is possibly still quite relevant to the OP's question about whether a higher sample rate makes for better digital audio.

 

Can we agree that the engineers who designed 8x oversampling into DAC chips decades ago did so for solid engineering reasons? And if that's granted, then can we go from there to saying there is no engineering reason for a decimation-interpolation sequence in the middle of the recording/playback chain, but this exists solely as an artifact of the way the music industry has evolved?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Well if PCM 48/24 (or 96/24) is enough, then bits and sampling at that are enough whether via some purist version that performs as it should or whether you have in between DSP. If we go straight to some higher rate and higher number of bits transmitted that sounds like an unengineered method that will work. An engineered method is more efficient.

 

That is before we mention recording, mixing, mastering, and processing being easily doable in PCM. Then upon playback digital volume control and Room EQ that sort of thing also being easily handled in PCM formats. If the cost is some measured inaudible improvement and simplicity at the cost of those things plus a higher bit thru put, doesn't sound like a clear cut great trade off to me.

 

There's a solid reason for that: it saves tremendous amounts of storage space and transmission bandwidth.

 

 

Wow, talk about making a virtue of necessity! The engineers who developed 8x oversampling within a very short time after digital audio was first marketed were not operating from an "unengineered method," they were engineering the solution that gave acceptable results at greatest cost and resource efficiency. 352.8/384KHz sampling rates are not "some purist version," but what was acceptable performance to non-audiophile engineers working for major consumer corporations trying to sell to the mass market decades ago.

 

Of course you're correct that currently at least engineering of recordings is more easily done in PCM. That's the 8x oversampling I was mentioning. So the "debate" would be between maintaining DXD (8x) rates all the way through and DSD conversion, which allows less expensive filtering (that's why delta-sigma has become ubiquitous, it's cheaper for what we'll call, without wishing to get into the old PCM-DSD debates, a substantially equivalent result).

 

I live in a semi-rural setting a few miles from a town of 4,000 people, and the download speed from my ISP (a cable company that is a relatively small local family owned operation) is 50mbps, soon to double to 100. I can buy a 4TB external HDD for a little over $100. When you talk about storage space and bandwidth, you're really reprising my argument that the RedBook standard is an artifact of history - digital audio was developed when storage space and bandwidth were much greater limitations than they are today for most people in a position to be concerned about things like the sonic impact of boutique power supplies.

 

It is very likely that the engineers who designed 8x oversampling into DAC chips decades ago did so for solid engineering reasons as engineers dislike doing things irrationally (tho are sometimes forced to do irrational things by their MBA Overlords).

 

But best practices decades ago may also not be best practices today.

 

 

Indeed they may not be best practices today. Do you think those practices would have evolved in the direction of *more restricted* capabilities than a couple of decades ago, or greater capabilities?

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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